1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
|
/*
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* sample format and channel layout conversion audio filter
*/
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
#include "libavresample/avresample.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"
typedef struct ResampleContext {
AVAudioResampleContext *avr;
int64_t next_pts;
} ResampleContext;
static av_cold void uninit(AVFilterContext *ctx)
{
ResampleContext *s = ctx->priv;
if (s->avr) {
avresample_close(s->avr);
avresample_free(&s->avr);
}
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
AVFilterFormats *out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
AVFilterFormats *in_samplerates = ff_all_samplerates();
AVFilterFormats *out_samplerates = ff_all_samplerates();
AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
AVFilterChannelLayouts *out_layouts = ff_all_channel_layouts();
ff_formats_ref(in_formats, &inlink->out_formats);
ff_formats_ref(out_formats, &outlink->in_formats);
ff_formats_ref(in_samplerates, &inlink->out_samplerates);
ff_formats_ref(out_samplerates, &outlink->in_samplerates);
ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AVFilterLink *inlink = ctx->inputs[0];
ResampleContext *s = ctx->priv;
char buf1[64], buf2[64];
int ret;
if (s->avr) {
avresample_close(s->avr);
avresample_free(&s->avr);
}
if (inlink->channel_layout == outlink->channel_layout &&
inlink->sample_rate == outlink->sample_rate &&
inlink->format == outlink->format)
return 0;
if (!(s->avr = avresample_alloc_context()))
return AVERROR(ENOMEM);
av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
/* if both the input and output formats are s16 or u8, use s16 as
the internal sample format */
if (av_get_bytes_per_sample(inlink->format) <= 2 &&
av_get_bytes_per_sample(outlink->format) <= 2)
av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);
if ((ret = avresample_open(s->avr)) < 0)
return ret;
outlink->time_base = (AVRational){ 1, outlink->sample_rate };
s->next_pts = AV_NOPTS_VALUE;
av_get_channel_layout_string(buf1, sizeof(buf1),
-1, inlink ->channel_layout);
av_get_channel_layout_string(buf2, sizeof(buf2),
-1, outlink->channel_layout);
av_log(ctx, AV_LOG_VERBOSE,
"fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
return 0;
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
ResampleContext *s = ctx->priv;
int ret = avfilter_request_frame(ctx->inputs[0]);
/* flush the lavr delay buffer */
if (ret == AVERROR_EOF && s->avr) {
AVFilterBufferRef *buf;
int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
outlink->sample_rate,
ctx->inputs[0]->sample_rate,
AV_ROUND_UP);
if (!nb_samples)
return ret;
buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
if (!buf)
return AVERROR(ENOMEM);
ret = avresample_convert(s->avr, (void**)buf->extended_data,
buf->linesize[0], nb_samples,
NULL, 0, 0);
if (ret <= 0) {
avfilter_unref_buffer(buf);
return (ret == 0) ? AVERROR_EOF : ret;
}
buf->pts = s->next_pts;
ff_filter_samples(outlink, buf);
return 0;
}
return ret;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
ResampleContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
if (s->avr) {
AVFilterBufferRef *buf_out;
int delay, nb_samples, ret;
/* maximum possible samples lavr can output */
delay = avresample_get_delay(s->avr);
nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
outlink->sample_rate, inlink->sample_rate,
AV_ROUND_UP);
buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
ret = avresample_convert(s->avr, (void**)buf_out->extended_data,
buf_out->linesize[0], nb_samples,
(void**)buf->extended_data, buf->linesize[0],
buf->audio->nb_samples);
av_assert0(!avresample_available(s->avr));
if (s->next_pts == AV_NOPTS_VALUE) {
if (buf->pts == AV_NOPTS_VALUE) {
av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
"assuming 0.\n");
s->next_pts = 0;
} else
s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
outlink->time_base);
}
if (ret > 0) {
buf_out->audio->nb_samples = ret;
if (buf->pts != AV_NOPTS_VALUE) {
buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
outlink->time_base) -
av_rescale(delay, outlink->sample_rate,
inlink->sample_rate);
} else
buf_out->pts = s->next_pts;
s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
ff_filter_samples(outlink, buf_out);
}
avfilter_unref_buffer(buf);
} else
ff_filter_samples(outlink, buf);
}
AVFilter avfilter_af_resample = {
.name = "resample",
.description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
.priv_size = sizeof(ResampleContext),
.uninit = uninit,
.query_formats = query_formats,
.inputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ },
{ .name = NULL}},
.outputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
.request_frame = request_frame },
{ .name = NULL}},
};
|